Session Initiation Protocol

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The Session Initiation Protocol (SIP) is a signaling protocol used for initiating, maintaining, and
terminating real-time sessions that include voice, video and messaging applications.[1] SIP is used
for signaling and controlling multimedia communication sessions in applications of Internet
telephony for voice and video calls, in private IP telephone systems, in instant
messaging over Internet Protocol(IP) networks as well as mobile phone calling over LTE (VoLTE).
The protocol defines the specific format of messages exchanged and the sequence of
communications for cooperation of the participants. SIP is a text-based protocol, incorporating many
elements of the Hypertext Transfer Protocol (HTTP) and the Simple Mail Transfer
Protocol (SMTP).[2] A call established with SIP may consist of multiple media streams, but no
 separate streams are required for applications, such as text messaging, that exchange data as
payload in the SIP message.
SIP works in conjunction with several other protocols that specify and carry the session media. Most
commonly, media type and parameter negotiation and media setup is performed with the Session
Description Protocol (SDP), which is carried as payload in SIP messages. SIP is designed to be
independent of the underlying transport layer protocol, and can be used with the User Datagram
Protocol (UDP), the Transmission Control Protocol (TCP), and the Stream Control Transmission
Protocol (SCTP). For secure transmissions of SIP messages over insecure network links, the
protocol may be encrypted with Transport Layer Security (TLS). For the transmission of media
streams (voice, video) the SDP payload carried in SIP messages typically employs the Real-time
Transport Protocol (RTP) or the Secure Real-time Transport Protocol (SRTP).

Contents

 1History
 2Protocol operation
 3Network elements
o 3.1User agent
o 3.2Proxy server
o 3.3Redirect server
o 3.4Registrar
o 3.5Session border controller
o 3.6Gateway
 4SIP messages
o 4.1Requests
o 4.2Responses
 5Transactions
 6Instant messaging and presence
 7Conformance testing
 8Performance testing
 9Applications
 10Implementations
 11SIP-ISUP interworking
 12Encryption
 13See also
 14References
 15Bibliography
 16External links

History[edit]
SIP was originally designed by Mark Handley, Henning Schulzrinne, Eve Schooler and Jonathan
Rosenberg in 1996. The protocol was standardized as RFC 2543 in 1999. In November 2000, SIP
was accepted as a 3GPP signaling protocol and permanent element of the IP Multimedia
Subsystem (IMS) architecture for IP-based streaming multimedia services in cellular networks. In
June 2002 the specification was revised in RFC 3261[3] and various extensions and clarifications
have been published since.[4]
SIP was designed to provide a signaling and call setup protocol for IP-based communications
supporting the call processing functions and features present in the public switched telephone
network (PSTN) with a vision of supporting new multimedia applications. It has been extended
for video conferencing, streaming media distribution, instant messaging, presence information, file
transfer, Internet fax and online games.[1][5][6]
SIP is distinguished by its proponents for having roots in the Internet community rather than in
the telecommunications industry. SIP has been standardized primarily by the IETF, while other
protocols, such as H.323, have traditionally been associated with the International
Telecommunication Union (ITU).

Protocol operation[edit]
SIP is only involved for the signaling operations of a media communication session and is primarily
used to set up and terminate voice or video calls. SIP can be used to establish two-party (unicast) or
multiparty (multicast) sessions. It also allows modification of existing calls. The modification can
involve changing addresses or ports, inviting more participants, and adding or deleting media
streams. SIP has also found applications in messaging applications, such as instant messaging, and
event subscription and notification.
SIP works in conjunction with several other protocols that specify the media format and coding and
that carry the media once the call is set up. For call setup, the body of a SIP message contains
a Session Description Protocol (SDP) data unit, which specifies the media format, codec and media
communication protocol. Voice and video media streams are typically carried between the terminals
using the Real-time Transport Protocol (RTP) or Secure Real-time Transport Protocol (SRTP).[2][7]
Every resource of a SIP network, such as user agents, call routers, and voicemail boxes, are
identified by a Uniform Resource Identifier (URI). The syntax of the URI follows the general standard
syntax also used in Web services and e-mail.[8] The URI scheme used for SIP is sip and a typical SIP
URI has the form sip:username@domainname or sip:username@hostport,
where domainname requires DNS SRV records to locate the servers for SIP domain
while hostport can be an IP address or a fully qualified domain name of the host and port. If secure
transmission is required, the scheme sips is used.[9][10]
SIP employs design elements similar to the HTTP request and response transaction model.[11] Each
transaction consists of a client request that invokes a particular method or function on the server and
at least one response. SIP reuses most of the header fields, encoding rules and status codes of
HTTP, providing a readable text-based format.
SIP can be carried by several transport layer protocols including Transmission Control
Protocol (TCP), User Datagram Protocol (UDP), and Stream Control Transmission
Protocol(SCTP).[12][13] SIP clients typically use TCP or UDP on port numbers 5060 or 5061 for SIP
traffic to servers and other endpoints. Port 5060 is commonly used for non-encrypted signaling traffic
whereas port 5061 is typically used for traffic encrypted with Transport Layer Security (TLS).
SIP-based telephony networks often implement call processing features of Signaling System
7 (SS7), for which special SIP protocol extensions exist, although the two protocols themselves are
very different. SS7 is a centralized protocol, characterized by a complex central network architecture
and dumb endpoints (traditional telephone handsets). SIP is a client-server protocol of equipotent
peers. SIP features are implemented in the communicating endpoints, while the traditional SS7
architecture is in use only between switching centers.

Network elements[edit]
The network elements that use the Session Initiation Protocol for communication are called SIP user
agents. Each user agent (UA) performs the function of a user agent client (UAC) when it is
requesting a service function, and that of a user agent server (UAS) when responding to a request.
Thus, any two SIP endpoints may in principle operate without any intervening SIP infrastructure.
However, for network operational reasons, for provisioning public services to users, and for directory
services, SIP defines several specific types of network server elements. Each of these service
elements also communicates within the client-server model implemented in user agent clients and
servers[14].
User agent[edit]
A user agent is a logical network end-point that sends or receives SIP messages and manages SIP
sessions. User agents have client and server components. The user agent client (UAC) sends SIP
requests. The user agent server (UAS) receives requests and returns a SIP response. Unlike other
network protocols that fix the roles of client and server, e.g., in HTTP, in which a web browser only
acts as a client, and never as a server, SIP requires both peers to implement both roles. The roles of
UAC and UAS only last for the duration of a SIP transaction.[5]
A SIP phone is an IP phone that implements client and server functions of a SIP user agent and
provides the traditional call functions of a telephone, such as dial, answer, reject, call hold, and call
transfer.[15][16] SIP phones may be implemented as a hardware device or as a softphone. As vendors
increasingly implement SIP as a standard telephony platform, the distinction between hardware-
based and software-based SIP phones is blurred and SIP elements are implemented in the basic
firmware functions of many IP-capable communications devices such as smartphones.
In SIP, as in HTTP, the user agent may identify itself using a message header field (User-Agent),
containing a text description of the software, hardware, or the product name. The user agent field is
sent in request messages, which means that the receiving SIP server can evaluate this information
to perform device-specific configuration or feature activation. Operators of SIP network elements
sometimes store this information in customer account portals,[17] where it can be useful in diagnosing
SIP compatibility problems or in the display of service status.
Proxy server[edit]
A proxy server is a network server with UAC and UAS components that functions as an intermediary
entity for the purpose of performing requests on behalf of other network elements. A proxy server
primarily plays the role of routing, meaning that its job is to ensure that a request is sent to another
entity closer to the targeted user. Proxies are also useful for enforcing policy, such as for
determining whether a user is allowed to make a call. A proxy interprets, and, if necessary, rewrites
specific parts of a request message before forwarding it.
SIP proxy servers that route messages to more than one destination are called forking proxies. The
forking of SIP requests means that multiple dialogs can be established from a single request. This
explains the need for the two-sided dialog identifier; without a contribution from the recipients, the
originator could not disambiguate the multiple dialogs established from a single request.
SIP forking refers to the process of "forking" a single SIP call to multiple SIP endpoints. This is a very
powerful feature of SIP. A single call can ring many endpoints at the same time. SIP forking allows a
desk phone ring at the same time as a mobile, allowing a call to be taken from either device.
Redirect server[edit]
A redirect server is a user agent server that generates 3xx (redirection) responses to requests it
receives, directing the client to contact an alternate set of URIs. A redirect server allows proxy
servers to direct SIP session invitations to external domains.
Registrar[edit]
SIP user agent registration to SIP registrar with authentication.

A registrar is a SIP endpoint that provides a location service. It accepts REGISTER requests,
recording the address and other parameters from the user agent. For subsequent requests it
provides an essential means to locate possible communication peers on the network. The location
service links one or more IP addresses to the SIP URI of the registering agent. Multiple user agents
may register for the same URI, with the result that all registered user agents receive the calls to the
URI.
SIP registrars are logical elements, and are often co-located with SIP proxies. To improve network
scalability, location services may instead be located with a redirect server.
Session border controller[edit]

Establishment of a session through a back-to-back user agent.

Session border controllers serve as middle boxes between user agents and SIP servers for various
types of functions, including network topology hiding and assistance in NAT traversal.
Gateway[edit]
Gateways can be used to interconnect a SIP network to other networks, such as the public switched
telephone network, which use different protocols or technologies.

SIP messages[edit]
SIP is a text-based protocol with syntax similar to that of HTTP. There are two different types of SIP
messages: requests and responses. The first line of a request has a method, defining the nature of
the request, and a Request-URI, indicating where the request should be sent.[18] The first line of a
response has a response code.
Requests[edit]
Requests initiate a functionality of the protocol. They are sent by a user agent client to the server,
and are answered with one or more SIP responses, which return a result code of the transaction,
and generally indicate the success, failure, or other state of the transaction.
 SIP requests

Request RFC
Description Notes
name references

Register the URI listed in the To-

header field with a location server and The command implements a location
REGISTER RFC 3261
associates it with the network address service.
given in a Contact header field.

Initiate a dialog for establishing a call. When sent during an established dialog
INVITE The request is sent by a user agent (reinvite) it modifies the sessions, for RFC 3261
client to a user agent server. example placing a call on hold.

Confirm that an entity has received a

ACK RFC 3261
final response to an INVITE request.

Signal termination of a dialog and end This message may be sent by either
BYE RFC 3261
a call. endpoint of a dialog.

Usually means terminating a call while

CANCEL Cancel any pending request. RFC 3261
it is still ringing, before answer.

Modify the state of a session without

UPDATE RFC 3311
changing the state of the dialog.

Ask recipient to issue a request for the

REFER RFC 3515
purpose of call transfer.

PRACK is sent in response to

PRACK Provisional acknowledgement. RFC 3262
provisional response (1xx).

Initiates a subscription for notification

SUBSCRIBE RFC 6665
of events from a notifier.
 Inform a subscriber of notifications of
NOTIFY RFC 6665
a new event.

Publish an event to a notification

PUBLISH RFC 3903
server.

Used in instant messaging

MESSAGE Deliver a text message. RFC 3428
applications.

Send mid-session information that This method is often used for DTMF
INFO RFC 6086
does not modify the session state. relay.

It is often used for

OPTIONS Query the capabilities of an endpoint. RFC 3261
NAT keepalive purposes.

Responses[edit]
Main article: List of SIP response codes
Responses are sent by the user agent server indicating the result of a received request. Several
classes of responses are recognized, determined by the numerical range of result codes:[19]

 1xx: Provisional responses to requests indicate the request was valid and is being processed.
 2xx: Successful completion of the request. As a response to an INVITE, it indicates a call is
established. The most common code is 200, which is an unqualified success report.
 3xx: Call redirection is needed for completion of the request. The request must be completed
with a new destination.
 4xx: The request cannot be completed at the server for a variety of reasons, including bad
request syntax (code 400).
 5xx: The server failed to fulfill an apparently valid request, including server internal errors (code
500).
 6xx: The request cannot be fulfilled at any server. It indicates a global failure, including call
rejection by the destination.

Transactions[edit]
Example: User1's UAC uses an invite client transaction to send the initial INVITE (1) message. If no response
is received after a timer controlled wait period the UAC may choose to terminate the transaction or retransmit
the INVITE. Once a response is received, User1 is confident the INVITE was delivered reliably. User1's UAC
must then acknowledge the response. On delivery of the ACK (2), both sides of the transaction are complete.
In this case, a dialog may have been established.[20]

SIP defines a transaction mechanism to control the exchanges between participants and deliver
messages reliably. A transaction is a state of a session, which is controlled by various timers. Client
transactions send requests and server transactions respond to those requests with one or more
responses. The responses may include provisional responses with a response code in the form 1xx,
and one or multiple final responses (2xx – 6xx).
Transactions are further categorized as either type invite or type non-invite. Invite transactions differ
in that they can establish a long-running conversation, referred to as a dialog in SIP, and so include
an acknowledgment (ACK) of any non-failing final response, e.g., 200 OK.

Instant messaging and presence[edit]

The Session Initiation Protocol for Instant Messaging and Presence Leveraging
Extensions (SIMPLE) is the SIP-based suite of standards for instant messaging and presence
information. Message Session Relay Protocol (MSRP) allows instant message sessions and file
transfer.

Conformance testing[edit]
The SIP developer community meets regularly at conferences organized by SIP Forum to test
interoperability of SIP implementations.[21] The TTCN-3 test specification language, developed by a
task force at ETSI (STF 196), is used for specifying conformance tests for SIP implementations.[22]

Performance testing[edit]
When developing SIP software or deploying a new SIP infrastructure, it is very important to test
capability of servers and IP networks to handle certain call load: number of concurrent calls and
number of calls per second. SIP performance tester software is used to simulate SIP and RTP traffic
to see if the server and IP network are stable under the call load.[23] The software measures
performance indicators like answer delay, answer/seizure ratio, RTP jitter and packet loss, round-trip
delay time.

Applications[edit]
SIP connection is a marketing term for voice over Internet Protocol (VoIP) services offered by
many Internet telephony service providers (ITSPs). The service provides routing of telephone calls
from a client's private branch exchange (PBX) telephone system to the public switched telephone
network (PSTN). Such services may simplify corporate information system infrastructure by
sharing Internet access for voice and data, and removing the cost for Basic Rate Interface (BRI)
or Primary Rate Interface (PRI) telephone circuits.
SIP trunking is a similar marketing term preferred for when the service is used to simplify a telecom
infrastructure by sharing the carrier access circuit for voice, data, and Internet traffic while removing
the need for Primary Rate Interface (PRI) circuits.[24][25]
SIP-enabled video surveillance cameras can initiate calls to alert the operator of events, such as
motion of objects in a protected area.
SIP is used in audio over IP for broadcasting applications where it provides an interoperable means
for audio interfaces from different manufacturers to make connections with one another.[26]

Implementations[edit]
The U.S. National Institute of Standards and Technology (NIST), Advanced Networking
Technologies Division provides a public-domain Java implementation[27] that serves as a reference
implementation for the standard. The implementation can work in proxy server or user agent
scenarios and has been used in numerous commercial and research projects. It
supports RFC 3261 in full and a number of extension RFCs including RFC 6665 (event notification)
and RFC 3262 (reliable provisional responses).
Numerous other commercial and open-source SIP implementations exist. See List of SIP software.

SIP-ISUP interworking[edit]
SIP-I, or the Session Initiation Protocol with encapsulated ISUP, is a protocol used to create, modify,
and terminate communication sessions based on ISUP using SIP and IP networks. Services using
SIP-I include voice, video telephony, fax and data. SIP-I and SIP-T[28] are two protocols with similar
features, notably to allow ISUP messages to be transported over SIP networks. This preserves all of
the detail available in the ISUP header, which is important as there are many country-specific
variants of ISUP that have been implemented over the last 30 years, and it is not always possible to
express all of the same detail using a native SIP message. SIP-I was defined by the ITU-T, whereas
SIP-T was defined via the IETF RFC route.[29]

Encryption[edit]
Concerns about the security of calls via the public Internet have been addressed by encryption of the
SIP protocol for secure transmission. The URI scheme sips is used to mandate that each hop over
which the request is forwarded up to the target domain must be secured with Transport Layer
Security (TLS). The last hop from the proxy of the target domain to the user agent has to be secured
according to local policies. TLS protects against attackers who try to listen on the signaling link but it
does not provide end-to-end security to prevent espionage and law enforcement interception, as the
encryption is only hop-by-hop and every single intermediate proxy has to be trusted.
End-to-end security may also be achieved with secure tunneling and IPsec, but most service
providers that offer secure connections use TLS for securing signaling.[citation needed]TLS connections use
URIs in the form sips:user@example.com. The media streams which are separate connections from
the signaling stream, may be encrypted with the Secure Real-time Transport Protocol (SRTP). The
key exchange for SRTP is performed with SDES (RFC 4568), or with ZRTP (RFC 6189). One may
also add a MIKEY (RFC 3830) exchange to SIP to determine session keys for use with SRTP.